gstreamer/gst/rtmp2/gstrtmp2src.c
Vivia Nikolaidou 2386858a91 Add files from gst-rtmp
For master, without autotools.
2019-11-05 13:52:55 +00:00

839 lines
24 KiB
C

/* GStreamer
* Copyright (C) 2014 David Schleef <ds@schleef.org>
* Copyright (C) 2017 Make.TV, Inc. <info@make.tv>
* Contact: Jan Alexander Steffens (heftig) <jsteffens@make.tv>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 51 Franklin Street, Suite 500,
* Boston, MA 02110-1335, USA.
*/
/**
* SECTION:element-gstrtmp2src
*
* The rtmp2src element receives input streams from an RTMP server.
*
* <refsect2>
* <title>Example launch line</title>
* |[
* gst-launch -v rtmp2src ! decodebin ! fakesink
* ]|
* FIXME Describe what the pipeline does.
* </refsect2>
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include "gstrtmp2src.h"
#include "gstrtmp2locationhandler.h"
#include "rtmp/rtmpclient.h"
#include "rtmp/rtmpmessage.h"
#include <gst/base/gstpushsrc.h>
#include <string.h>
GST_DEBUG_CATEGORY_STATIC (gst_rtmp2_src_debug_category);
#define GST_CAT_DEFAULT gst_rtmp2_src_debug_category
/* prototypes */
#define GST_RTMP2_SRC(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTMP2_SRC,GstRtmp2Src))
#define GST_IS_RTMP2_SRC(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTMP2_SRC))
typedef struct
{
GstPushSrc parent_instance;
/* properties */
GstRtmpLocation location;
gboolean async_connect;
/* stuff */
gboolean running, flushing;
GMutex lock;
GCond cond;
GstTask *task;
GRecMutex task_lock;
GMainLoop *loop;
GMainContext *context;
GCancellable *cancellable;
GstRtmpConnection *connection;
guint32 stream_id;
GstBuffer *message;
gboolean sent_header;
GstClockTime last_ts;
} GstRtmp2Src;
typedef struct
{
GstPushSrcClass parent_class;
} GstRtmp2SrcClass;
/* GObject virtual functions */
static void gst_rtmp2_src_set_property (GObject * object,
guint property_id, const GValue * value, GParamSpec * pspec);
static void gst_rtmp2_src_get_property (GObject * object,
guint property_id, GValue * value, GParamSpec * pspec);
static void gst_rtmp2_src_finalize (GObject * object);
static void gst_rtmp2_src_uri_handler_init (GstURIHandlerInterface * iface);
/* GstBaseSrc virtual functions */
static gboolean gst_rtmp2_src_start (GstBaseSrc * src);
static gboolean gst_rtmp2_src_stop (GstBaseSrc * src);
static gboolean gst_rtmp2_src_unlock (GstBaseSrc * src);
static gboolean gst_rtmp2_src_unlock_stop (GstBaseSrc * src);
static GstFlowReturn gst_rtmp2_src_create (GstBaseSrc * src, guint64 offset,
guint size, GstBuffer ** outbuf);
/* Internal API */
static void gst_rtmp2_src_task_func (gpointer user_data);
static void client_connect_done (GObject * source, GAsyncResult * result,
gpointer user_data);
static void start_play_done (GObject * object, GAsyncResult * result,
gpointer user_data);
static void connect_task_done (GObject * object, GAsyncResult * result,
gpointer user_data);
enum
{
PROP_0,
PROP_LOCATION,
PROP_SCHEME,
PROP_HOST,
PROP_PORT,
PROP_APPLICATION,
PROP_STREAM,
PROP_SECURE_TOKEN,
PROP_USERNAME,
PROP_PASSWORD,
PROP_AUTHMOD,
PROP_TIMEOUT,
PROP_TLS_VALIDATION_FLAGS,
PROP_ASYNC_CONNECT,
};
/* pad templates */
static GstStaticPadTemplate gst_rtmp2_src_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("video/x-flv")
);
/* class initialization */
G_DEFINE_TYPE_WITH_CODE (GstRtmp2Src, gst_rtmp2_src, GST_TYPE_PUSH_SRC,
G_IMPLEMENT_INTERFACE (GST_TYPE_URI_HANDLER,
gst_rtmp2_src_uri_handler_init);
G_IMPLEMENT_INTERFACE (GST_TYPE_RTMP_LOCATION_HANDLER, NULL));
static void
gst_rtmp2_src_class_init (GstRtmp2SrcClass * klass)
{
GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
GstBaseSrcClass *base_src_class = GST_BASE_SRC_CLASS (klass);
gst_element_class_add_static_pad_template (GST_ELEMENT_CLASS (klass),
&gst_rtmp2_src_src_template);
gst_element_class_set_static_metadata (GST_ELEMENT_CLASS (klass),
"RTMP source element", "Source", "Source element for RTMP streams",
"Make.TV, Inc. <info@make.tv>");
gobject_class->set_property = gst_rtmp2_src_set_property;
gobject_class->get_property = gst_rtmp2_src_get_property;
gobject_class->finalize = gst_rtmp2_src_finalize;
base_src_class->start = GST_DEBUG_FUNCPTR (gst_rtmp2_src_start);
base_src_class->stop = GST_DEBUG_FUNCPTR (gst_rtmp2_src_stop);
base_src_class->unlock = GST_DEBUG_FUNCPTR (gst_rtmp2_src_unlock);
base_src_class->unlock_stop = GST_DEBUG_FUNCPTR (gst_rtmp2_src_unlock_stop);
base_src_class->create = GST_DEBUG_FUNCPTR (gst_rtmp2_src_create);
g_object_class_override_property (gobject_class, PROP_LOCATION, "location");
g_object_class_override_property (gobject_class, PROP_SCHEME, "scheme");
g_object_class_override_property (gobject_class, PROP_HOST, "host");
g_object_class_override_property (gobject_class, PROP_PORT, "port");
g_object_class_override_property (gobject_class, PROP_APPLICATION,
"application");
g_object_class_override_property (gobject_class, PROP_STREAM, "stream");
g_object_class_override_property (gobject_class, PROP_SECURE_TOKEN,
"secure-token");
g_object_class_override_property (gobject_class, PROP_USERNAME, "username");
g_object_class_override_property (gobject_class, PROP_PASSWORD, "password");
g_object_class_override_property (gobject_class, PROP_AUTHMOD, "authmod");
g_object_class_override_property (gobject_class, PROP_TIMEOUT, "timeout");
g_object_class_override_property (gobject_class, PROP_TLS_VALIDATION_FLAGS,
"tls-validation-flags");
g_object_class_install_property (gobject_class, PROP_ASYNC_CONNECT,
g_param_spec_boolean ("async-connect", "Async connect",
"Connect on READY, otherwise on first push", TRUE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
GST_DEBUG_CATEGORY_INIT (gst_rtmp2_src_debug_category, "rtmp2src", 0,
"debug category for rtmp2src element");
}
static void
gst_rtmp2_src_init (GstRtmp2Src * self)
{
self->async_connect = TRUE;
g_mutex_init (&self->lock);
g_cond_init (&self->cond);
self->task = gst_task_new (gst_rtmp2_src_task_func, self, NULL);
g_rec_mutex_init (&self->task_lock);
gst_task_set_lock (self->task, &self->task_lock);
}
static void
gst_rtmp2_src_uri_handler_init (GstURIHandlerInterface * iface)
{
gst_rtmp_location_handler_implement_uri_handler (iface, GST_URI_SRC);
}
static void
gst_rtmp2_src_set_property (GObject * object, guint property_id,
const GValue * value, GParamSpec * pspec)
{
GstRtmp2Src *self = GST_RTMP2_SRC (object);
switch (property_id) {
case PROP_LOCATION:
gst_rtmp_location_handler_set_uri (GST_RTMP_LOCATION_HANDLER (self),
g_value_get_string (value));
break;
case PROP_SCHEME:
GST_OBJECT_LOCK (self);
self->location.scheme = g_value_get_enum (value);
GST_OBJECT_UNLOCK (self);
break;
case PROP_HOST:
GST_OBJECT_LOCK (self);
g_free (self->location.host);
self->location.host = g_value_dup_string (value);
GST_OBJECT_UNLOCK (self);
break;
case PROP_PORT:
GST_OBJECT_LOCK (self);
self->location.port = g_value_get_int (value);
GST_OBJECT_UNLOCK (self);
break;
case PROP_APPLICATION:
GST_OBJECT_LOCK (self);
g_free (self->location.application);
self->location.application = g_value_dup_string (value);
GST_OBJECT_UNLOCK (self);
break;
case PROP_STREAM:
GST_OBJECT_LOCK (self);
g_free (self->location.stream);
self->location.stream = g_value_dup_string (value);
GST_OBJECT_UNLOCK (self);
break;
case PROP_SECURE_TOKEN:
GST_OBJECT_LOCK (self);
g_free (self->location.secure_token);
self->location.secure_token = g_value_dup_string (value);
GST_OBJECT_UNLOCK (self);
break;
case PROP_USERNAME:
GST_OBJECT_LOCK (self);
g_free (self->location.username);
self->location.username = g_value_dup_string (value);
GST_OBJECT_UNLOCK (self);
break;
case PROP_PASSWORD:
GST_OBJECT_LOCK (self);
g_free (self->location.password);
self->location.password = g_value_dup_string (value);
GST_OBJECT_UNLOCK (self);
break;
case PROP_AUTHMOD:
GST_OBJECT_LOCK (self);
self->location.authmod = g_value_get_enum (value);
GST_OBJECT_UNLOCK (self);
break;
case PROP_TIMEOUT:
GST_OBJECT_LOCK (self);
self->location.timeout = g_value_get_uint (value);
GST_OBJECT_UNLOCK (self);
break;
case PROP_TLS_VALIDATION_FLAGS:
GST_OBJECT_LOCK (self);
self->location.tls_flags = g_value_get_flags (value);
GST_OBJECT_UNLOCK (self);
break;
case PROP_ASYNC_CONNECT:
GST_OBJECT_LOCK (self);
self->async_connect = g_value_get_boolean (value);
GST_OBJECT_UNLOCK (self);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, property_id, pspec);
break;
}
}
static void
gst_rtmp2_src_get_property (GObject * object, guint property_id,
GValue * value, GParamSpec * pspec)
{
GstRtmp2Src *self = GST_RTMP2_SRC (object);
switch (property_id) {
case PROP_LOCATION:
GST_OBJECT_LOCK (self);
g_value_take_string (value, gst_rtmp_location_get_string (&self->location,
TRUE));
GST_OBJECT_UNLOCK (self);
break;
case PROP_SCHEME:
GST_OBJECT_LOCK (self);
g_value_set_enum (value, self->location.scheme);
GST_OBJECT_UNLOCK (self);
break;
case PROP_HOST:
GST_OBJECT_LOCK (self);
g_value_set_string (value, self->location.host);
GST_OBJECT_UNLOCK (self);
break;
case PROP_PORT:
GST_OBJECT_LOCK (self);
g_value_set_int (value, self->location.port);
GST_OBJECT_UNLOCK (self);
break;
case PROP_APPLICATION:
GST_OBJECT_LOCK (self);
g_value_set_string (value, self->location.application);
GST_OBJECT_UNLOCK (self);
break;
case PROP_STREAM:
GST_OBJECT_LOCK (self);
g_value_set_string (value, self->location.stream);
GST_OBJECT_UNLOCK (self);
break;
case PROP_SECURE_TOKEN:
GST_OBJECT_LOCK (self);
g_value_set_string (value, self->location.secure_token);
GST_OBJECT_UNLOCK (self);
break;
case PROP_USERNAME:
GST_OBJECT_LOCK (self);
g_value_set_string (value, self->location.username);
GST_OBJECT_UNLOCK (self);
break;
case PROP_PASSWORD:
GST_OBJECT_LOCK (self);
g_value_set_string (value, self->location.password);
GST_OBJECT_UNLOCK (self);
break;
case PROP_AUTHMOD:
GST_OBJECT_LOCK (self);
g_value_set_enum (value, self->location.authmod);
GST_OBJECT_UNLOCK (self);
break;
case PROP_TIMEOUT:
GST_OBJECT_LOCK (self);
g_value_set_uint (value, self->location.timeout);
GST_OBJECT_UNLOCK (self);
break;
case PROP_TLS_VALIDATION_FLAGS:
GST_OBJECT_LOCK (self);
g_value_set_flags (value, self->location.tls_flags);
GST_OBJECT_UNLOCK (self);
break;
case PROP_ASYNC_CONNECT:
GST_OBJECT_LOCK (self);
g_value_set_boolean (value, self->async_connect);
GST_OBJECT_UNLOCK (self);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, property_id, pspec);
break;
}
}
static void
gst_rtmp2_src_finalize (GObject * object)
{
GstRtmp2Src *self = GST_RTMP2_SRC (object);
gst_buffer_replace (&self->message, NULL);
g_clear_object (&self->cancellable);
g_clear_object (&self->connection);
g_clear_object (&self->task);
g_rec_mutex_clear (&self->task_lock);
g_mutex_clear (&self->lock);
g_cond_clear (&self->cond);
gst_rtmp_location_clear (&self->location);
G_OBJECT_CLASS (gst_rtmp2_src_parent_class)->finalize (object);
}
static gboolean
gst_rtmp2_src_start (GstBaseSrc * src)
{
GstRtmp2Src *self = GST_RTMP2_SRC (src);
gboolean async;
GST_OBJECT_LOCK (self);
async = self->async_connect;
GST_OBJECT_UNLOCK (self);
GST_INFO_OBJECT (self, "Starting (%s)", async ? "async" : "delayed");
g_clear_object (&self->cancellable);
self->running = TRUE;
self->cancellable = g_cancellable_new ();
self->stream_id = 0;
self->sent_header = FALSE;
self->last_ts = GST_CLOCK_TIME_NONE;
if (async) {
gst_task_start (self->task);
}
return TRUE;
}
static gboolean
quit_invoker (gpointer user_data)
{
g_main_loop_quit (user_data);
return G_SOURCE_REMOVE;
}
static void
stop_task (GstRtmp2Src * self)
{
gst_task_stop (self->task);
self->running = FALSE;
if (self->cancellable) {
GST_DEBUG_OBJECT (self, "Cancelling");
g_cancellable_cancel (self->cancellable);
}
if (self->loop) {
GST_DEBUG_OBJECT (self, "Stopping loop");
g_main_context_invoke_full (self->context, G_PRIORITY_DEFAULT_IDLE,
quit_invoker, g_main_loop_ref (self->loop),
(GDestroyNotify) g_main_loop_unref);
}
g_cond_broadcast (&self->cond);
}
static gboolean
gst_rtmp2_src_stop (GstBaseSrc * src)
{
GstRtmp2Src *self = GST_RTMP2_SRC (src);
GST_DEBUG_OBJECT (self, "stop");
g_mutex_lock (&self->lock);
stop_task (self);
g_mutex_unlock (&self->lock);
gst_task_join (self->task);
return TRUE;
}
static gboolean
gst_rtmp2_src_unlock (GstBaseSrc * src)
{
GstRtmp2Src *self = GST_RTMP2_SRC (src);
GST_DEBUG_OBJECT (self, "unlock");
g_mutex_lock (&self->lock);
self->flushing = TRUE;
g_cond_broadcast (&self->cond);
g_mutex_unlock (&self->lock);
return TRUE;
}
static gboolean
gst_rtmp2_src_unlock_stop (GstBaseSrc * src)
{
GstRtmp2Src *self = GST_RTMP2_SRC (src);
GST_DEBUG_OBJECT (self, "unlock_stop");
g_mutex_lock (&self->lock);
self->flushing = FALSE;
g_mutex_unlock (&self->lock);
return TRUE;
}
static GstFlowReturn
gst_rtmp2_src_create (GstBaseSrc * src, guint64 offset, guint size,
GstBuffer ** outbuf)
{
GstRtmp2Src *self = GST_RTMP2_SRC (src);
GstBuffer *message, *buffer;
GstRtmpMeta *meta;
guint32 timestamp = 0;
static const guint8 flv_header_data[] = {
0x46, 0x4c, 0x56, 0x01, 0x01, 0x00, 0x00, 0x00,
0x09, 0x00, 0x00, 0x00, 0x00,
};
GST_LOG_OBJECT (self, "create");
g_mutex_lock (&self->lock);
if (self->running) {
gst_task_start (self->task);
}
while (!self->message) {
if (!self->running) {
g_mutex_unlock (&self->lock);
return GST_FLOW_EOS;
}
if (self->flushing) {
g_mutex_unlock (&self->lock);
return GST_FLOW_FLUSHING;
}
g_cond_wait (&self->cond, &self->lock);
}
message = self->message;
self->message = NULL;
g_cond_signal (&self->cond);
g_mutex_unlock (&self->lock);
meta = gst_buffer_get_rtmp_meta (message);
if (!meta) {
GST_ELEMENT_ERROR (self, CORE, FAILED,
("Internal error: No RTMP meta on buffer"),
("No RTMP meta on %" GST_PTR_FORMAT, message));
gst_buffer_unref (message);
return GST_FLOW_ERROR;
}
if (GST_BUFFER_DTS_IS_VALID (message)) {
GstClockTime last_ts = self->last_ts, ts = GST_BUFFER_DTS (message);
if (GST_CLOCK_TIME_IS_VALID (last_ts) && last_ts > ts) {
GST_LOG_OBJECT (self, "Timestamp regression: %" GST_TIME_FORMAT
" > %" GST_TIME_FORMAT, GST_TIME_ARGS (last_ts), GST_TIME_ARGS (ts));
}
self->last_ts = ts;
timestamp = ts / GST_MSECOND;
}
buffer = gst_buffer_copy_region (message, GST_BUFFER_COPY_MEMORY, 0, -1);
{
guint8 *tag_header = g_malloc (11);
GstMemory *memory =
gst_memory_new_wrapped (0, tag_header, 11, 0, 11, tag_header, g_free);
GST_WRITE_UINT8 (tag_header, meta->type);
GST_WRITE_UINT24_BE (tag_header + 1, meta->size);
GST_WRITE_UINT24_BE (tag_header + 4, timestamp);
GST_WRITE_UINT8 (tag_header + 7, timestamp >> 24);
GST_WRITE_UINT24_BE (tag_header + 8, 0);
gst_buffer_prepend_memory (buffer, memory);
}
{
guint8 *tag_footer = g_malloc (4);
GstMemory *memory =
gst_memory_new_wrapped (0, tag_footer, 4, 0, 4, tag_footer, g_free);
GST_WRITE_UINT32_BE (tag_footer, meta->size + 11);
gst_buffer_append_memory (buffer, memory);
}
if (!self->sent_header) {
GstMemory *memory = gst_memory_new_wrapped (GST_MEMORY_FLAG_READONLY,
(guint8 *) flv_header_data, sizeof flv_header_data, 0,
sizeof flv_header_data, NULL, NULL);
gst_buffer_prepend_memory (buffer, memory);
self->sent_header = TRUE;
}
*outbuf = buffer;
gst_buffer_unref (message);
return GST_FLOW_OK;
}
/* Mainloop task */
static void
gst_rtmp2_src_task_func (gpointer user_data)
{
GstRtmp2Src *self = GST_RTMP2_SRC (user_data);
GMainContext *context;
GMainLoop *loop;
GTask *connector;
GST_DEBUG_OBJECT (self, "gst_rtmp2_src_task starting");
g_mutex_lock (&self->lock);
context = self->context = g_main_context_new ();
g_main_context_push_thread_default (context);
loop = self->loop = g_main_loop_new (context, TRUE);
connector = g_task_new (self, self->cancellable, connect_task_done, NULL);
GST_OBJECT_LOCK (self);
gst_rtmp_client_connect_async (&self->location, self->cancellable,
client_connect_done, connector);
GST_OBJECT_UNLOCK (self);
g_mutex_unlock (&self->lock);
g_main_loop_run (loop);
g_mutex_lock (&self->lock);
g_clear_pointer (&self->loop, g_main_loop_unref);
g_clear_pointer (&self->connection, gst_rtmp_connection_close_and_unref);
g_cond_broadcast (&self->cond);
g_mutex_unlock (&self->lock);
while (g_main_context_pending (context)) {
GST_DEBUG_OBJECT (self, "iterating main context to clean up");
g_main_context_iteration (context, FALSE);
}
g_main_context_pop_thread_default (context);
g_mutex_lock (&self->lock);
g_clear_pointer (&self->context, g_main_context_unref);
gst_buffer_replace (&self->message, NULL);
g_mutex_unlock (&self->lock);
GST_DEBUG_OBJECT (self, "gst_rtmp2_src_task exiting");
}
static void
client_connect_done (GObject * source, GAsyncResult * result,
gpointer user_data)
{
GTask *task = user_data;
GstRtmp2Src *self = g_task_get_source_object (task);
GError *error = NULL;
GstRtmpConnection *connection;
connection = gst_rtmp_client_connect_finish (result, &error);
if (!connection) {
g_task_return_error (task, error);
g_object_unref (task);
return;
}
g_task_set_task_data (task, connection, g_object_unref);
if (g_task_return_error_if_cancelled (task)) {
g_object_unref (task);
return;
}
GST_OBJECT_LOCK (self);
gst_rtmp_client_start_play_async (connection, self->location.stream,
g_task_get_cancellable (task), start_play_done, task);
GST_OBJECT_UNLOCK (self);
}
static void
start_play_done (GObject * source, GAsyncResult * result, gpointer user_data)
{
GTask *task = G_TASK (user_data);
GstRtmp2Src *self = g_task_get_source_object (task);
GstRtmpConnection *connection = g_task_get_task_data (task);
GError *error = NULL;
if (g_task_return_error_if_cancelled (task)) {
g_object_unref (task);
return;
}
if (gst_rtmp_client_start_play_finish (connection, result,
&self->stream_id, &error)) {
g_task_return_pointer (task, g_object_ref (connection),
gst_rtmp_connection_close_and_unref);
} else {
g_task_return_error (task, error);
}
g_task_set_task_data (task, NULL, NULL);
g_object_unref (task);
}
static void
got_message (GstRtmpConnection * connection, GstBuffer * buffer,
gpointer user_data)
{
GstRtmp2Src *self = GST_RTMP2_SRC (user_data);
GstRtmpMeta *meta = gst_buffer_get_rtmp_meta (buffer);
guint32 min_size = 1;
g_return_if_fail (meta);
if (meta->mstream != self->stream_id) {
GST_DEBUG_OBJECT (self, "Ignoring %s message with stream %" G_GUINT32_FORMAT
" != %" G_GUINT32_FORMAT, gst_rtmp_message_type_get_nick (meta->type),
meta->mstream, self->stream_id);
return;
}
switch (meta->type) {
case GST_RTMP_MESSAGE_TYPE_VIDEO:
min_size = 6;
break;
case GST_RTMP_MESSAGE_TYPE_AUDIO:
min_size = 2;
break;
case GST_RTMP_MESSAGE_TYPE_DATA_AMF0:
break;
default:
GST_DEBUG_OBJECT (self, "Ignoring %s message, wrong type",
gst_rtmp_message_type_get_nick (meta->type));
return;
}
if (meta->size < min_size) {
GST_DEBUG_OBJECT (self, "Ignoring too small %s message (%" G_GUINT32_FORMAT
" < %" G_GUINT32_FORMAT ")",
gst_rtmp_message_type_get_nick (meta->type), meta->size, min_size);
return;
}
g_mutex_lock (&self->lock);
while (self->message) {
if (!self->running) {
goto out;
}
g_cond_wait (&self->cond, &self->lock);
}
self->message = gst_buffer_ref (buffer);
g_cond_signal (&self->cond);
out:
g_mutex_unlock (&self->lock);
return;
}
static void
error_callback (GstRtmpConnection * connection, GstRtmp2Src * self)
{
g_mutex_lock (&self->lock);
if (self->cancellable) {
g_cancellable_cancel (self->cancellable);
} else if (self->loop) {
GST_INFO_OBJECT (self, "Connection error");
stop_task (self);
}
g_mutex_unlock (&self->lock);
}
static void
control_callback (GstRtmpConnection * connection, gint uc_type,
guint stream_id, GstRtmp2Src * self)
{
GST_INFO_OBJECT (self, "stream %u got %s", stream_id,
gst_rtmp_user_control_type_get_nick (uc_type));
if (uc_type == GST_RTMP_USER_CONTROL_TYPE_STREAM_EOF && stream_id == 1) {
GST_INFO_OBJECT (self, "went EOS");
stop_task (self);
}
}
static void
send_connect_error (GstRtmp2Src * self, GError * error)
{
if (!error) {
GST_ERROR_OBJECT (self, "Connect failed with NULL error");
GST_ELEMENT_ERROR (self, RESOURCE, FAILED, ("Failed to connect"), (NULL));
return;
}
if (g_error_matches (error, G_IO_ERROR, G_IO_ERROR_CANCELLED)) {
GST_DEBUG_OBJECT (self, "Connection was cancelled (%s)",
GST_STR_NULL (error->message));
return;
}
GST_ERROR_OBJECT (self, "Failed to connect (%s:%d): %s",
g_quark_to_string (error->domain), error->code,
GST_STR_NULL (error->message));
if (g_error_matches (error, G_IO_ERROR, G_IO_ERROR_PERMISSION_DENIED)) {
GST_ELEMENT_ERROR (self, RESOURCE, NOT_AUTHORIZED,
("Not authorized to connect"), ("%s", GST_STR_NULL (error->message)));
} else if (g_error_matches (error, G_IO_ERROR, G_IO_ERROR_CONNECTION_REFUSED)) {
GST_ELEMENT_ERROR (self, RESOURCE, OPEN_READ,
("Could not connect"), ("%s", GST_STR_NULL (error->message)));
} else {
GST_ELEMENT_ERROR (self, RESOURCE, FAILED,
("Failed to connect"),
("error %s:%d: %s", g_quark_to_string (error->domain), error->code,
GST_STR_NULL (error->message)));
}
}
static void
connect_task_done (GObject * object, GAsyncResult * result, gpointer user_data)
{
GstRtmp2Src *self = GST_RTMP2_SRC (object);
GTask *task = G_TASK (result);
GError *error = NULL;
g_mutex_lock (&self->lock);
g_warn_if_fail (g_task_is_valid (task, object));
if (self->cancellable == g_task_get_cancellable (task)) {
g_clear_object (&self->cancellable);
}
self->connection = g_task_propagate_pointer (task, &error);
if (self->connection) {
gst_rtmp_connection_set_input_handler (self->connection,
got_message, g_object_ref (self), g_object_unref);
g_signal_connect_object (self->connection, "error",
G_CALLBACK (error_callback), self, 0);
g_signal_connect_object (self->connection, "stream-control",
G_CALLBACK (control_callback), self, 0);
} else {
send_connect_error (self, error);
stop_task (self);
g_error_free (error);
}
g_cond_broadcast (&self->cond);
g_mutex_unlock (&self->lock);
}